Issues 19000 - 19999

[..]
ASTERISK-19000: Multiple ChanSpy users cannot hear each other
ASTERISK-19002: Using realtime sippeers: no value given for outbound proxy on line 0 of sip.conf
ASTERISK-19003: asterisk closes TLS connection after receiving ACK for 401 following INVITE sent by Snom m9
ASTERISK-19004: RTP timeout won't work with locally bridged SIP channels
ASTERISK-19005: Not working parameter mailcmd in voicemail.conf
ASTERISK-19009: Deadlock on sip_new and load_realtime_queue.
ASTERISK-19011: crashing res_odbc because of use of obj->con while reconnecting
ASTERISK-19012: CLONE - [patch] CCSS: Sending a NOTIFY without the Subscription-State header
ASTERISK-19013: T.38 port negotiation problem
ASTERISK-19029: amaflags is not copied to channel for outgoing sip call
ASTERISK-19030: Invalid host= declaration causes crash
ASTERISK-19031: Asterisk can seg fault on invalid tcptls_session reference
ASTERISK-19034: CLONE -[patch] New manager option enabledevents
ASTERISK-19039: Indirect Ring Group Routing Not Ringing All Phones in Ring Group
ASTERISK-19040: Asterisk 1.8.9.0 Blockers
ASTERISK-19041: Asterisk 10.1.0 Blockers
ASTERISK-19042: When joining ConfBridge, channel mutex can be free'd more times then it is locked
ASTERISK-19048: T.38 Fallback to G.711 fails upon 503 response
ASTERISK-19049: CDR wasn't generated after doing redirect through AGI
ASTERISK-19050: Wrong transport for outgoing INVITE
ASTERISK-19053: Investigate why the gateway_mix2 test is failing.
ASTERISK-19055: Memory leaks in app_followme find_realtime
ASTERISK-19056: Incorrect description for MESSAGE_SEND_STATUS variable in main.message.c
ASTERISK-19057: [patch] message-summary NOTIFY: Port in Message-Account added twice and mwi_from (sip.conf) has no effect
ASTERISK-19058: Messagesend and SIPFROMUSER
ASTERISK-19061: Wiki Documentation on Realtime Database Connector possible incorrect syntax for extconfig.conf file.
ASTERISK-19062: app_stack: cannot access memory at address 0x0
ASTERISK-19063: The channel fall on pickup with option sendrpid=yes
ASTERISK-19079: Asterisk will not build under Freebsd with GCC 4.6 installed
ASTERISK-19081: Call files in /var/spool/asterisk/outgoing are sometime not read and processed by pbx_spool.c
ASTERISK-19082: Forwarding voicemail generate error in multi-tenant configuration
ASTERISK-19087: CLONE -core show channels randomly shows IP instead of IAX account
ASTERISK-19088: CLONE -Implicit Assumption About Dynamic Features
ASTERISK-19089: faxdetect=yes in sip.conf general overrides faxdetect=no in peer configuration
ASTERISK-19091: ConfBridge cannot handle multiple menu DTMF selections in rapid succession
ASTERISK-19092: cisco phone 79xx can't register
ASTERISK-19093: sip reload not loading all users
ASTERISK-19094: Incorrect -x command line parameter behavior
ASTERISK-19095: REGRESSION after r336294: Music on hold when sip call is put on hold doesnt work anymore after 1.8.8.0-rc1
ASTERISK-19096: Allow specifying which MixMonitor to stop
ASTERISK-19097: Click To Call Busy Destination Results In Hangup
ASTERISK-19098: chan_mgcp being enabled without dependencies met
ASTERISK-19099: ConfBridge, set marked ( to send video) from console.
ASTERISK-19100: ConfBridge crashes on closing timer when destroying conference bridge
ASTERISK-19103: When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
ASTERISK-19104: transmit_refer doesn't send "Replaces" in tag "Refer-To:"
ASTERISK-19105: [JABBER± Jitsi call cause Asterisk to SEGFAULT
ASTERISK-19106: SIP registration fails after temporary dns failure
ASTERISK-19107: #include configuration file if it exists (do not fail if it's missing)
ASTERISK-19108: TESTTIME function replies errors
ASTERISK-19109: [patch] "deaf" participant support in ConfBridge
ASTERISK-19126: Cannot install asterisk on mac os lion
ASTERISK-19127: Asterisk does not quit on SIGTERM
ASTERISK-19128: Asterisk 1.8.10 Blockers
ASTERISK-19129: Asterisk 10.2.0 Blockers
ASTERISK-19133: Memory leak using asterisk T.38 to/from T.30 gateway
ASTERISK-19135: Asterisk ended with exit status 134
ASTERISK-19136: crash in odbc when using app_voicemail
ASTERISK-19137: patten matching wrong when issue "dialplan reload"
ASTERISK-19138: CLI does not Honor key bindings or keybinding file ~/.editrc
ASTERISK-19139: SIP regression in 1.8 branch
ASTERISK-19140: Put DAHDISpan and DAHDIChannel on some AMI events
ASTERISK-19141: pub_lua Extensions and Execution of CONNECTED_LINE_CALLER_SEND_MACRO
ASTERISK-19142: manager parameter channelvars=CHANNEL(dahdi_span) causes segmentation fault on Hangup
ASTERISK-19143: Core dump when adding dialplan extension
ASTERISK-19153: [patch] - Sms sender is not parsed correctly in incoming sms
ASTERISK-19154: huge number of sip OPTION on 'sip reload'
ASTERISK-19155: Memory leak in app_voicemail.c when using IMAP
ASTERISK-19156: 1.8 SVN: channel.c:1474 __ast_queue_frame: Exceptionally long voice queue length queuing to Local during paging
ASTERISK-19157: Failed to authenticate on INVITE to '"Anonymous"
ASTERISK-19159: Asterisk fails to start MOH when SDP specifies connection IP of 0.0.0.0 only
ASTERISK-19161: [patch] Add function REGISTRANT() that retrieves the peer that auto-registered an extension
ASTERISK-19163: Got SIP response 400 "Bad Request" after Hangup
ASTERISK-19164: ForkCDR with 'e' option to set end time is overzealous
ASTERISK-19165: Empty CDR userfield and wrong UniqueID values stored in Master.csv when using Originate from AMI
ASTERISK-19166: Retransmitted REGISTER requests are rejected with 401 (stale=true).
ASTERISK-19167: Fix skipped tests in Asterisk Test Suite
ASTERISK-19169: [patch] CallerID send before ring problem detected in chain_dahdi.c
ASTERISK-19170: realtime queues fail to load queue information when there arent valid queue_members in the queue_members table
ASTERISK-19171: sip tcp fails with secret
ASTERISK-19172: Inconstistency for realtime colmn lastms
ASTERISK-19173: All blind transfers failing on 1.8.9.0-rc1
ASTERISK-19176: The 'w' modifier support for ISDN spans was lost when sig_pri.c was extracted from chan_dahdi.c. Dial(DAHDI/g0/1234w888)
ASTERISK-19178: Asterisk 10 beta2 disconnects on reload
ASTERISK-19179: RTP inactivity SIP / ooh323 wont work
ASTERISK-19180: ast_cel_fabricate_channel_from_event causes AMI VarSet events to be sent for a temporary/dummy channel
ASTERISK-19181: SIP-Provider without "SIP/2.0 180 Ringing" no Audio when connected to DAHDi
ASTERISK-19182: Crash in ast_channel_get_full() with partial name
ASTERISK-19183: (Sporadically) missing connectedline event to caller channel in directed pickup app
ASTERISK-19184: Crash at attempt to attended transfer a call
ASTERISK-19186: Func_CURL is missing from wiki.asterisk.org, 1.8 functions section
ASTERISK-19188: asterisk crashes if there no confbridge-join file
ASTERISK-19189: AEL Macro and AELSub functions do not pass EXTEN variable, breaking CDR destination field
ASTERISK-19190: AJAM Digest missing session cookie
ASTERISK-19191: In AMI Redirect action somtimes channels get hangup while redirecting channels to meet me room
ASTERISK-19192: ERROR we couldn't allocate a port for RTP instance
ASTERISK-19193: Asterisk ended with exit status 134
ASTERISK-19196: Queue and local channels - Agent hunting order incorrect
ASTERISK-19197: Calls from VOIP to Dahdi requiring transcoding fail
ASTERISK-19198: Parallel make jobs break build
ASTERISK-19199: Neither MATH nor $[] expression have the ABS(X) (absolute value)
ASTERISK-19200: Not work alwaysauthreject=yes
ASTERISK-19201: TLS Manager Bind Port - random port - not configurable
ASTERISK-19202: CSipSimple (trunk) crushes Asterisk 1.8.8.1 (openSuse)
ASTERISK-19203: Resource leak in SIP/TCP
ASTERISK-19204: Manager API opens on random port on reload, TLS address not loaded as set
ASTERISK-19205: Most Unique pattern matching broken when trailing "-" is part of extension
ASTERISK-19206: Segmentation fault: menuselect/nmenuselect menuselect.makeopts
ASTERISK-19209: Attended transfer failes
ASTERISK-19213: deadlock on bultin atxfer
ASTERISK-19215: Segfault chan_sip originating call
ASTERISK-19216: cdr_pgsql reload failure
ASTERISK-19220: chan_sip deadlock
ASTERISK-19221: asterisk process hangs
ASTERISK-19222: dialplan add extension documentation issue
ASTERISK-19223: Called party keeps ringing until calling party has send a cancel
ASTERISK-19231: Abort signal 6 raises when using 'sip show peers' with realtime peers
ASTERISK-19232: Notifycid sending -1 instead of 1
ASTERISK-19233: patch to fix inband DTMF in chan_ooh323
ASTERISK-19234: Asterisk changes "From" header to "asterisk" when CALLERID(num-pres)=prohib_passed_screen is set
ASTERISK-19235: confbridge fails: chan_sip.c:6544 sip_write: Can't send 10 type frames with SIP write
ASTERISK-19240: UnParkedCall event does not contain the related parking lot name
ASTERISK-19241: Cannot compile with Embedded Modules
ASTERISK-19242: AMI QUEUESTATUS not working correctly in 10.0.1 (CLI queue show not working correct as well)
ASTERISK-19243: DEVICE_STATE not correct when in h extension
ASTERISK-19244: g729 not offered in SIP INVITE
ASTERISK-19245: Fresh install of 1.8.9.0-rc3 hangs during module load and pegs one processor at 100%
ASTERISK-19246: possible bug: Audiohook flag values overlap
ASTERISK-19247: Delaying destroy of SIP INVITE dialog fail while call is allready Bridged.
ASTERISK-19249: AMI PauseMonitor or UnpauseMonitor With Missing or Unknown Channel Forcibly Disconnects AMI Session.
ASTERISK-19250: --enable-dev-mode should also apply to editline
ASTERISK-19251: Manager eventq fills up with events with Usecount neq 0
ASTERISK-19252: qualify for h323
ASTERISK-19254: When working in real time with ARA and MySQL the backslashes not works properly
ASTERISK-19264: ASTERISK-19202 creates: trap invalid opcode ip:516aa2 sp:7fff224a4640 error:0 in asterisk[400000+196000] on x86_64 builds
ASTERISK-19265: ESwitch not Converting Variables
ASTERISK-19266: flood of SQL warnings on 1.8 - 10.1 upgrade
ASTERISK-19267: RSA key for TLS should not be stored in same file as cert
ASTERISK-19268: Need to specify TLS peer verification policy per-peer
ASTERISK-19270: CallerID missing on local channel
ASTERISK-19271: Asterisk 1.8.11.0 Blockers
ASTERISK-19272: Asterisk 10.3.0 Blockers
ASTERISK-19273: Store Asterisk Manager HTTP Sessions in persistent storage.
ASTERISK-19276: Google Calendar periodic event miss updating
ASTERISK-19277: [patch]endlessly repeating error: "poll failed: Bad file descriptor"
ASTERISK-19279: Asterisk stops processing Local channels - CLI is full of messages "Exceptionally long voice queue length queuing to Local/XXX"
ASTERISK-19281: "sip show peers" show incorrect columns
ASTERISK-19282: Add F option to Bridge command
ASTERISK-19283: Add F option to Queue command (transfer on hangup)
ASTERISK-19285: [regression] Deadlock in asterisk 1.8.9.0 (possible chan_agent and queues interaction)
ASTERISK-19289: chan_iax2.so: undefined symbol: ast_aes_set_encrypt_key
ASTERISK-19290: Voicemailmain password not recognized from Aastra 480i phone in versions past 10.0.1
ASTERISK-19291: Background in realtime
ASTERISK-19292: New "dialplan remove context" and modification of "dialplan add include"
ASTERISK-19293: Got SIP response 400 "Missing Subscription-State header"
ASTERISK-19294: Asterisk 1.8.6.0 failed to switch RTP destination when receiving a SIP reinvite
ASTERISK-19295: Segfault on "sip show peers" on Solaris
ASTERISK-19296: Attended transfer and hangup events
ASTERISK-19297: Call from 'SIPX' to extension 'xxxxxxxxx' rejected because extension not found in context 'from-SIPX'
ASTERISK-19298: segmentation fault in chan_ooh323
ASTERISK-19299: [patch] AgentLogin Option To Skip Password Prompt
ASTERISK-19300: chan_skype can not load under 1.8.9.0
ASTERISK-19301: ooh323 trunk to AVAYA
ASTERISK-19302: Messagesend
ASTERISK-19303: Asterisk does not acknowledge the ACK received to terminate the dialog.
ASTERISK-19304: New feature to send udptl packets directly between both call legs
ASTERISK-19305: After reciving INVITE with FROM user without phone number asterisk crashes with segfault
ASTERISK-19306: Invalid parameters in rt_handle_member (app_queue.c: create_queue_member: No location at interface '')
ASTERISK-19307: When a jabber server is configured as type=component, asterisk crashes with segmentation fault.
ASTERISK-19308: problem with transit calls ooh323-dahdi(pri)-panasonic 500
ASTERISK-19309: [patch] DUNDi message routing bug
ASTERISK-19310: 'i' option is defined twice at AST_APP_OPTIONS macro in app_page.c
ASTERISK-19311: ParkAndAnnounce crash asterisk
ASTERISK-19312: No DTMF decoding on outbound call via SS7 E1 channel
ASTERISK-19313: [patch] incorrect handling of UPDATE response with canreinvite=update
ASTERISK-19315: Impossibly High Lagged Value (Asterisk 1.8.8.1)
ASTERISK-19316: Asterisk cannot detect canceled calls on analog lines
ASTERISK-19317: QueueLog does not log ringnoanswer if the caller abandons while the agent's extension is ringing
ASTERISK-19318: Asterisk locks up during Page cmd
ASTERISK-19319: [patch] Triggers dialplan actions when specific CONTROL_FRAMES are detected on a channel.
ASTERISK-19320: SIGSEGV when starting within mgcp module
ASTERISK-19321: Transfer application ignores port information
ASTERISK-19322: Polycom blind SIP transfer to park extension plays parking orbit number prompt to caller extension after transfer.
ASTERISK-19332: chan_usbradio fails to compile under Ubuntu natty (--enable-dev-mode)
ASTERISK-19334: Adaptive CDR via ODBC driver can't handle UTF8-type fields in database
ASTERISK-19335: MeetMeAdmin(confno,N) mutes admins
ASTERISK-19336: h exten is not run in the context that calls a AEL macro
ASTERISK-19337: app_voicemail fails to compile with imap storage
ASTERISK-19340: [patch] CALLERID(subaddr) only allows ASCII
ASTERISK-19341: Missing initialization on bind_addr
ASTERISK-19342: Initialize parking hints with NOT_INUSE state
ASTERISK-19346: Registering multiple Google calendars via caldav crashes Asterisk
ASTERISK-19347: File descriptor errors from res_timing_timerfd
ASTERISK-19348: With alwaysauthreject=yes AND allowguest=no Asterisk fails to report a SIP Security Event
ASTERISK-19350: [Regression] Asterisk realtime via ODBC - null is inserted in brackets '()'
ASTERISK-19351: MeetMe does not record, WARNING: file.c:1168 ast_writefile: No such format ''
ASTERISK-19352: SIP warning message if only UDP is eanbled
ASTERISK-19353: musiconhold of remote party is replaced by local moh
ASTERISK-19354: ConfBridge does not close channel when using local channels
ASTERISK-19355: Call transfer with consultation frequently fails in cross-linked asterisk scenario (directmedia & sendrpid active)
ASTERISK-19356: Deadlock in cel_sqlite3_custom module reload
ASTERISK-19358: Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2
ASTERISK-19360: When using DialPlan in RealTime, TIMEOUT(response) does not work
ASTERISK-19361: Asterisk exited on signal 6: Related to sip show peers?
ASTERISK-19362: Loud squelch after conf-hasjoin: "is now in the conference" announcement
ASTERISK-19363: DAHDI PRI channel becomes unusable after a while, and calls fail over it.
ASTERISK-19364: add concise option to core show hints
ASTERISK-19365: Remote SIP Call legs are frequently not released in a cross-linked Asterisk scenario (directmedia & sendrpid)
ASTERISK-19366: Periodic RTCP receiver reports in cross-linked asterisk scenario although asterisk is no longer in the RTP path (directmedia)
ASTERISK-19367: Update Debian Install Prerequisite install
ASTERISK-19368: Queue penalty only work when QUEUE_MIN_PENALTY == QUEUE_MAX_PENALTY
ASTERISK-19369: SIP timeout to peer causes hangup, call does not continue to next priority.
ASTERISK-19370: format_ogg_vorbis fail to compile
ASTERISK-19371: Incorrect matching with new pattern match engine enabled
ASTERISK-19372: BUSY/INCOMPLETE/CONGESTION indications not passed to SS7 channel
ASTERISK-19373: Segmentation Fault in ast_udptl_write() due to bad memcpy() call
ASTERISK-19374: No audio in Gtalk calls
ASTERISK-19379: IAX channel chooses the wrong password for authentication
ASTERISK-19380: Asterisk 10.2.0-rc2 MessageSend() application reply to sender issue
ASTERISK-19381: Local channel don't inherited language
ASTERISK-19382: Park() ignores 'r' option, plays default MOH instead.
ASTERISK-19383: Asterisk 1.8.5.0 - atxfer authorization problem when a call returns for reject or no answer
ASTERISK-19384: REGRESSION - CLONE - CDR(accountcode) not accessable to 'Local' channels
ASTERISK-19385: "Callerid:" in call-files and Asterisk Manager doesn't work
ASTERISK-19386: Channel group not released after a channel has been Chanspy'd upon
ASTERISK-19387: Seg Fault upon Asterisk Startup
ASTERISK-19388: Make it possible to put any connected call on hold, not just bridged ones
ASTERISK-19389: Sending ACK in CANCLE dialog to the wrong destination
ASTERISK-19391: Unable to edit From Caller Name with RPID
ASTERISK-19397: Fix cause code for no channel available
ASTERISK-19407: Set CDR variable ignored on record created after after ForkCDR
ASTERISK-19409: TestSuite: twisted reactor incompatible with python subprocess module
ASTERISK-19411: Conference Participants are placed back on hold when marked user quits
ASTERISK-19416: H323 trunking failure.
ASTERISK-19417: CLONE - Unable to edit From Caller Name with RPID
ASTERISK-19418: Silence Suppression with TDM cards
ASTERISK-19419: Asterisk does not compile under dev mode with gcc 4.6.3
ASTERISK-19420: Segfault in chan_mgcp
ASTERISK-19421: app_rpt cannot be compiled with --enable-dev-mode (ubuntu 11.10+)
ASTERISK-19422: CCSS does not function if "sip" is used instead of "SIP" when dialing
ASTERISK-19423: Issue regarding CDR_ADAPTIVE_ODBC.c versus CDR_ODBC.c
ASTERISK-19424: Spurious hangups during ringing on analog DAHDI channels
ASTERISK-19425: Calls not released after BYE
ASTERISK-19426: Mixmonitor does not create file and record anything
ASTERISK-19428: Confbridge allows more max_members than set
ASTERISK-19429: ERROR[23303]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL when login to AMI
ASTERISK-19430: 1.8.9.1 SIP NOTIFY crashes 2wire (U-Verse) routers
ASTERISK-19431: Asterisk Russian language support missing voicemail prompts
ASTERISK-19432: Seg Fault in libresample upon Asterisk Startup
ASTERISK-19433: CLONE - Called party keeps ringing until calling party has send a cancel
ASTERISK-19434: Segmentation fault on starting ConfBridge
ASTERISK-19435: Asterisk segfaults in app_alarmreceiver
ASTERISK-19436: outbound fax over t38 gateway can't pass
ASTERISK-19440: ConfBridge does not turns off MOH when participant kicked out
ASTERISK-19441: menuselect makes <use> tags shown and executes as "Depends on:"
ASTERISK-19442: unaccepted attend transfer hangup caller
ASTERISK-19443: realtime peers are not loaded during start
ASTERISK-19444: Usage for CLI command 'devstate change' is truncated by an unnecessary comma
ASTERISK-19445: Incorrect values are specified as length in memcpy and memset
ASTERISK-19446: Improvement to MWI (with Teksavvy Tektalk service (Metaswitch Networks equipment)
ASTERISK-19447: [patch] Add IPv6 Address Support To Security Events Framework
ASTERISK-19448: ConfBridge crashes Asterisk when no timing module loaded.
ASTERISK-19449: Can't connect to asterisk's console using command asterist -r
ASTERISK-19450: there always an extra byte added to contact header while send "ACK" request.
ASTERISK-19451: va_start/va_copy and va_end do not always match up
ASTERISK-19452: ChanSpy with MixMonitor test sporadically fails
ASTERISK-19454: outbound proxy not being cleared which sip reload performed
ASTERISK-19455: SIP channels permanently stuck in system after BYE message received
ASTERISK-19456: Turn Off Warning Message When Bind Address Is Set To ANY
ASTERISK-19457: Re-add macro option for stdexten to support legacy dialplans
ASTERISK-19459: Asterisk sending BYE on wrong NIC
ASTERISK-19460: [patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string
ASTERISK-19461: ChanSpy - Improper refcounts avoid channel release
ASTERISK-19462: asterisk Illegal Instruction (core dumped)
ASTERISK-19463: Asterisk deadlocks during startup with mutex errors
ASTERISK-19465: P-Asserted-Identity Privacy
ASTERISK-19466: NSGate 39xx series gateway will not register with pedantic=yes
ASTERISK-19467: Error should always be logged if SIP message fails compliance check
ASTERISK-19468: Crash seems to be related to sip fax calls
ASTERISK-19469: Execution of audio playbacks from call time limits (Dial application 'L' option) do not reflect configured options
ASTERISK-19470: Documentation on app_amd is incorrect
ASTERISK-19471: ConfBridge does not record anything
ASTERISK-19483: Hangup during Read() or Record() causes hangup extension failure
ASTERISK-19487: AMI module reload causes deadlock
ASTERISK-19488: Rejected supervised transfer hangs up on calling party
ASTERISK-19491: MeetMe fails to remove participant from conference when user leaves during playback of 'conf-onlyperson.ulaw'
ASTERISK-19492: Group write permission removed from existing directory /etc/asterisk/. when updating
ASTERISK-19493: ChanSpy onto a Local channel can leave a hung channel
ASTERISK-19494: Blind Transfer to Park extension does not set timeout destination correctly on 2nd Park attempt.
ASTERISK-19495: Create CEL tests for the Asterisk Test Suite
ASTERISK-19497: ConfBridge recording does not work reliably
ASTERISK-19498: All calls via IAX2 fail
ASTERISK-19499: ConfBridge MOH is not working for transferee after attended transfer
ASTERISK-19500: T.38 session fails with '488 Not acceptable here' if within 5 seconds on asterisk 1.8
ASTERISK-19501: Channel group on a Local channel not released after it has been Chanspy'd upon
ASTERISK-19502: Wrong port specified on SIP INVITE response when using custom TCP port
ASTERISK-19503: Aastra 480i loses mwi light after every reboot until reloading Asterisk.
ASTERISK-19504: Queue Memeber Penalty not considered if queue reloaded after penalty modifieing
ASTERISK-19505: Crash during high usage when Dial Time out set to 280
ASTERISK-19506: LIMIT_WARNING_FILE plays warning to both participants BUT one after another, not at the same time
ASTERISK-19508: res_srtp.so crash with snom phone 370 on srtp_unprotect_rtcp
ASTERISK-19510: RTP stream/Directmedia/Musiconhold/SIP not working
ASTERISK-19511: Dial I option ignored if dial forked and one fork redirects
ASTERISK-19512: force rport inconsistent between sip show peer and peers
ASTERISK-19513: app_voicemail fails to compile with IMAP storage
ASTERISK-19514: Queue realtime does not remove members after removed from database: Queue fraud has no realtime members defined. No need for update
ASTERISK-19515: Need hooks for resource to leverage for NAT hole poking for media streams
ASTERISK-19516: [patch] Enable RFC 4662/Broadsoft Resource list subscriptions in Asterisk 1.8.9.0
ASTERISK-19518: Voice Mail with IMAP storage reports erroneous error message in FreePBX configurations
ASTERISK-19519: [patch] valid IP address in RTP offer when Asterisk is attached to several networks
ASTERISK-19520: Call IAX2 - PRI strong distortion when using 16kHz speex
ASTERISK-19521: chan_iax2 does not honor trunkfreq config option
ASTERISK-19522: realtime peers are not loaded during start
ASTERISK-19531: Realtime SIP peers that explicitly unregister have incorrect device state.
ASTERISK-19532: Asterisk crashed after connecting with jabber server in component mode
ASTERISK-19533: Script run from #exec can't connect to the manager on initial start or restart of Asterisk
ASTERISK-19535: Dial/queue should handle HOLD/UNHOLD control frame similar to connected line updates.
ASTERISK-19536: Queue option ringinuse is ignored
ASTERISK-19537: Deadlock potential in ast_do_masquerade() because it calls ast_indicate with the channel lock held.
ASTERISK-19538: Asterisk segfaults on sippeers realtime redundancy
ASTERISK-19539: jabber outgoing messages become incomings
ASTERISK-19540: Use of GNU old-style field designator extension
ASTERISK-19541: Security Vulnerability: remotely exploitable stack overrun in Milliwatt
ASTERISK-19542: Security Vulnerability: remotely exploitable stack overflow in main/utils ast_parse_digest
ASTERISK-19547: utils.c:1236 ast_careful_fwrite: fflush() returned error: Bad file descriptor
ASTERISK-19548: Ability to run dialplan on callee channel before making call upon Dial()
ASTERISK-19549: Channel Hangup Handlers
ASTERISK-19550: Segfault in ast_readaudio_callback
ASTERISK-19551: Dial with Gosub autoservice error message is misleading
ASTERISK-19552: CDR logs on call transfers prints only last leg
ASTERISK-19553: Cannot perform feature attended transfer after pickup when using PickupChan
ASTERISK-19554: chan_unistim notes warnings about retransmissions of ACK and multiple ACKs received
ASTERISK-19555: SIP name in RDNIS, not CallerID number
ASTERISK-19556: Asteriskt thread use 99% cpu
ASTERISK-19557: [Regression] Segfault in res_jabber.c
ASTERISK-19558: SIP INVITE header is broken
ASTERISK-19559: No sound in calls after 1-2 seconds (SIP to IAX2)
ASTERISK-19560: broken CANCEL is record routes are added to ringing
ASTERISK-19561: libwat support (Wireless AT Library)
ASTERISK-19562: [patch] ConfBridge - Inconsistent hold-music behaviour
ASTERISK-19565: Investigate failures of the nat_supertest in the Asterisk Test Suite
ASTERISK-19567: Investigate module load / unload failures in dynamic_modules test in Asterisk Test Suite
ASTERISK-19571: [patch][feature] ConfBridge - Support for playing back arbitrary messages to individuals or the whole bridge
ASTERISK-19572: Proposed change to CDR MySQL table structure
ASTERISK-19573: hangup is not detected during call to func_curl within pbx_lua when a SIP caller hangs up
ASTERISK-19574: Directory application should set variable upon successful search.
ASTERISK-19575: AMI Hangup channels by regex
ASTERISK-19576: DTMF Passed Unreliably from DAHDI Analog to GTalk
ASTERISK-19577: Overcoming 64 callgroup / pickupgroup limit by creating "group contexts"
ASTERISK-19578: ERROR we couldn't allocate a port for RTP instance while DAHDI bridgeing
ASTERISK-19579: ERROR we couldn't allocate a port for RTP instance while DAHDI bridgeing
ASTERISK-19580: chan_gtalk crash Asterisk on outgoing calls
ASTERISK-19590: call completed the agent fixes a completecaller
ASTERISK-19591: CallCompletion atumatic cancel on call return
ASTERISK-19592: Security Vulnerability: heap overflow exists in chan_skinny's handling of KEYPAD_BUTTON_MESSAGE
ASTERISK-19594: app_meetme unable to write frame (stuck channel)
ASTERISK-19595: Inefficient wav49 disk writes
ASTERISK-19596: Memory Leak of 8gigs RAM.
ASTERISK-19597: Failure to pass NULL data pointer with AST_CONTROL_HOLD frame causes crash when MOH is started
ASTERISK-19598: Garbled audio using Page app and MulticastRTP channel
ASTERISK-19599: Announce parking slot number to callee
ASTERISK-19601: Failure of domain matching on authentication of INVITE request produces misleading NOTICE message; bypasses alwaysauthreject logic
ASTERISK-19603: Asterisk AMI truncates long responses over medium latency connections
ASTERISK-19604: pin parameter of meetme ignored for dynamically added conferences
ASTERISK-19606: CLONE - Directory application should set variable upon successful search.
ASTERISK-19608: Asterisk-1.8.x starts rejecting calls with cause code 44 after some time.
ASTERISK-19609: SRTP to RTP bridging with two crypto lines in SDP does not work
ASTERISK-19610: dsp.c can no longer detect a quick DTMF sequence
ASTERISK-19611: SIP stack stops working (deadlock?) if a call to a snom phone is redirected by "302 Moved temporarily" to chan_local
ASTERISK-19612: Asterisk crash with compiler flag LOW_MEMORY
ASTERISK-19613: Multibytes characters in files are not handled properly (signed char compared to int will get incorrect result if this byte is one of multibyte character)
ASTERISK-19614: ILBC 20ms force
ASTERISK-19618: Asterisk 1.8.12.0 Blockers
ASTERISK-19619: Asterisk 10.4.0 Blockers
ASTERISK-19620: directrtpsetup is not working anymore as expected/as in earlier Asterisk versions
ASTERISK-19628: Crash during blind transfer with chan_unistim
ASTERISK-19632: Trouble with fax gateway
ASTERISK-19633: Having any h extension in peer's context breaks unaccepted attended feature transfers
ASTERISK-19634: Sending DTMF tones using the AMI through the agent proxy channel doesnt work as expected
ASTERISK-19635: Hangup is always recorded in queue_log as COMPLETECALLER when 'h' extension is present
ASTERISK-19636: Asterisk crashes during attended transfer due to bad data pointer passed in HOLD frame from chan_iax2
ASTERISK-19637: Userfield variable cannot be correctly set in h extension
ASTERISK-19638: Problem: Asterisk wont start
ASTERISK-19639: [patch] - Deadlock in queue with attended transfer
ASTERISK-19640: aastra-xml UDP problem!
ASTERISK-19641: ConfBridge app plays conf-placeintoconf message to bridge, and not to joining channel
ASTERISK-19642: SIP channels permanently stuck in system after BYE message received
ASTERISK-19643: codec_dahdi: Block on frameout if the hardware has enough samples to complete a frame.
ASTERISK-19644: Loss of audio while running ChanSpy() on a specific prefix; channel gets stuck permanently.
ASTERISK-19645: Asterisk doesn't respect the video codec order
ASTERISK-19646: Fix typo \n in chan_sip SDP negotiation warning message
ASTERISK-19647: talktime 0 sec after transfer queueA->QueueB
ASTERISK-19651: Coverity Report: Fix issues for error type SIZEOF_MISMATCH
ASTERISK-19655: Coverity Report: Fix issues for error type NEGATIVE_RETURNS
ASTERISK-19662: Coverity Report: Fix issues for error type MISSING_BREAK
ASTERISK-19665: Coverity Report: Fix issues for error type RESOURCE_LEAK
ASTERISK-19668: Coverity Report: Fix issues for error type OVERRUN_STATIC
ASTERISK-19671: Coverity Report: Fix issues for error type REVERSE_NEGATIVE
ASTERISK-19675: MWI stops working after lifting handset or using phone
ASTERISK-19676: Fax Sessions are not always reported by 'fax show sessions'
ASTERISK-19677: SIP dial string //IPorHost does not work like expected
ASTERISK-19678: Manager disconnects on return of large dataset with action_command
ASTERISK-19680: Monitor application docs are missing/incorrect
ASTERISK-19682: Parsing of XML document tag <variable> malforms wiki documentation
ASTERISK-19684: [T.38 gateway] [irroot t38gateway-1.8 branch] I can receive faxes with t38 gateway, but send fails
ASTERISK-19708: Call Deflection with DAHDISendCallreroutingFacility on EuroISDN not working
ASTERISK-19709: fix so connectab takes the ser_dialog_custom table into account
ASTERISK-19711: Crash emanating from add_exten_to_pattern_tree()
ASTERISK-19712: Retrieve of fields from Calendar EWS
ASTERISK-19713: Asterisk segfaults on invalid datastore in channel destructor
ASTERISK-19714: No hungup after BYE.
ASTERISK-19716: Don't validate Contact URI hostpart when nat=yes
ASTERISK-19717: Attended transfer hangup
ASTERISK-19718: ast_app_inboxcount2() calls ast_inboxcount2_func without checking if it's assigned (instead checks ast_inboxcount_func)
ASTERISK-19719: Asterisk doesn't add Contact field in 200 OK when ACK for 401 response is out of order
ASTERISK-19720: Using SIP realtime with caching incoming calls are routed to dialplan context specified in sip.conf general context and not the context specified in SIP peer context until peer is refreshed from database
ASTERISK-19721: Asterisk core sounds, italian version
ASTERISK-19722: Manager disconnects under high originate load volume
ASTERISK-19723: Blind parking does not work anymore
ASTERISK-19724: [patch][feature] ConfBridge - Support for enabling MoH playback in conferences
ASTERISK-19725: [patch][feature] ConfBridge - Support for getting the name-recording file created when participants announce themselves
ASTERISK-19726: [patch][bug] ConfBridge - Users listening to MoH, and who should be muted, are often unmuted and recorded
ASTERISK-19727: MixMonitor does not work on local channels
ASTERISK-19730: Stack overflow in chan_sip when destroying mwipvt
ASTERISK-19734: Having any h extension in peer's context breaks unaccepted attended feature transfers
ASTERISK-19738: Calendar EWS does not attempt to extract the Body element in a CalendarItem and populate the description event field
ASTERISK-19748: Add LinkedID to AMI Events
ASTERISK-19749: Calendar EWS can't force event trigger to execute dialplan context without adding channel to calendar.conf
ASTERISK-19750: display_send=name_initial doesn't work
ASTERISK-19751: asterisk fails to autoconfigure - cannot find sqlite3
ASTERISK-19752: Add a channel variable/configuration option for defining pre-recorded announcement when "announce_join_leave=yes"
ASTERISK-19753: App Macro argument processing does not honor escaped, quoted, or nested parentheses commas.
ASTERISK-19754: Deadlock in chan_sip / pthread_timing
ASTERISK-19755: __ao2_ref() validates user_data twice
ASTERISK-19756: Use of asprintf with MALLOC_DEBUG could corrupt memory or crash.
ASTERISK-19758: main/asterisk.c rawmemchr() undefined on OpenBSD
ASTERISK-19759: Missing Payload Type From Events API
ASTERISK-19760: Update Security Events Unit Tests
ASTERISK-19761: mp3_read crash
ASTERISK-19762: Segfault in ast_frdup when invalid data length specified in duplicated frame
ASTERISK-19763: Confbridge with SIP to Local channel results in hung video; Exceptionally long queue length queuing to Bridge
ASTERISK-19764: Infinite loop with autoservice when looking for nonexistant extension label.
ASTERISK-19765: CLONE - ACK is ignored upon call-pickup with sendrpid=yes
ASTERISK-19766: Linkedid field is too short
ASTERISK-19767: dead lock in res_config_pgsl while dialing SIP extension
ASTERISK-19768: ConfBridge - sound_muted/sound_unmuted aren't playing
ASTERISK-19769: main/asterisk.c rawmemchr() undefined on *BSD ()
ASTERISK-19770: Security Vulnerability: Segmentation fault when receiving an out-of-dialogue SIP UPDATE including a rpid info
ASTERISK-19771: User is unable to customize sound_leader_has_left
ASTERISK-19772: [branch] Making it possible to set minimum DTMF duration without patching channel.c
ASTERISK-19773: Asterisk crash on issuing Asterisk-CLI 'reload' command multiple times on cli_aliases
ASTERISK-19775: Backgrace generation in Asterisk causes a seg fault
ASTERISK-19778: dialplan function MANAGER_ONLINE(username)
ASTERISK-19779: Asterisk segfaults when handling sip_security_event and attempting to load realtime peer with no realtime backend
ASTERISK-19780: Asterisk segfaults on invalid datastore in channel destructor
ASTERISK-19793: Only last realtime member of a queue is not actually removed from queue when removed from database
ASTERISK-19794: IAX channel won't hangup after musconhold stopped
ASTERISK-19795: Pinequeue: Play queue prompts in the background
ASTERISK-19796: race between ast_readaudio_callback and ast_closestream
ASTERISK-19797: GET_HEADER(Proxy-Authorization) get nothing
ASTERISK-19799: Apparent deadlock between ODBC Queue log and ODBC CDR
ASTERISK-19801: Deadlock with masquerade and chan_iax
ASTERISK-19802: Fails to set WRITE mode correctly, skips first file to be played when format is ULAW.
ASTERISK-19803: asterisk cli no tab completion for removing queue member with MEMBER_NAME variable
ASTERISK-19805: Race condition exists between hanging up Conference Bridge Channel and servicing the channel
ASTERISK-19807: Create masquerade "super-test" for Asterisk Test Suite
ASTERISK-19810: Asterisk 10: AGI 'record file' broken in SVN 364536
ASTERISK-19815: Crash in core show locks when BETTER_BACKTRACES is enabled
ASTERISK-19817: Call recording stops when call is transferred
ASTERISK-19818: Rework Asterisk Test Suite version parsing
ASTERISK-19820: wrapuptime is intermittently disregarded for queue calls
ASTERISK-19821: DTMF conversion SIP INFO to RFC2833 changes duration
ASTERISK-19822: EWS Calendar Integration Problem
ASTERISK-19825: Bridge stuck for several minutes after hangup - possible hangup control frame skip
ASTERISK-19827: Asterisk crash, whenever mwi => pass:user:authuser@host:port/mailbox is set in sip.conf
ASTERISK-19828: MESSAGE_SEND_STATUS set to SUCCESS despite response of 400 Bad Request
ASTERISK-19829: MessageSend disregards the port when specified in the from option of MessageSend(to[,from])
ASTERISK-19830: Asterisk receives an SDP offer with "recvonly" for video media but Asterisk responds with "sendrecv"
ASTERISK-19834: Memory leak caused by thread created by bridge_channel_join being neither joined nor detached
ASTERISK-19835: use webrtc iILBC code for codec_iLBC
ASTERISK-19836: Since change 325816 (T.38-gateway code) ReceiveFAX via T.38 immediately fail. always
ASTERISK-19837: Asterisk crashing regularly in 1.8.11.1 due to memory corruption
ASTERISK-19838: From Header has capital A in userpart Anonymous if CALLERID(pres)=unavailable, RFC uses lower case anonymous
ASTERISK-19839: Not all hints are displaying status correctly
ASTERISK-19840: Disable global atxfernoanswertimeout
ASTERISK-19842: POTS flashhook transfer causes deadlock
ASTERISK-19844: Broadvoice Got SIP response 503 Service Unavailable
ASTERISK-19845: Unable to register to sip through external pbx. Call ID Changed
ASTERISK-19846: sip users/peers not matched on incoming invite when there are multiple A records in DNS
ASTERISK-19847: Allow ConfBridge actions to be executed on channels in a conference via AMI
ASTERISK-19848: Deadlock in Asterisk in ast_parse_device_state / Dial
ASTERISK-19849: Issue also occurs when 3 people using IAX2 join the same meetme conference. Error messages recur until asterisk is restarted
ASTERISK-19851: Asterisk Crashes in chan_sip when failing to create ast_str in init_resp
ASTERISK-19852: Call pickup does not work with notifycid=yes
ASTERISK-19853: CDR(custom_field) field set in a DYNAMIC_FEATURE is reverted to the previous value when the call is terminated.
ASTERISK-19854: freeze channels showing in core show channels
ASTERISK-19856: Transfer is being denied when global allowtransfer=no, ignoring peer setting
ASTERISK-19857: Explore directmedia re-INVITE improvements between multiple Asterisk instances
ASTERISK-19859: cid_tag is not set according to the sip configuration anymore if get_rpid() != 0
ASTERISK-19860: Wrong CDR duration values because of dependency to CDR write time especially in cdr batch mode
ASTERISK-19861: External MWI subscriptions: Asterisk not responding to auth request
ASTERISK-19862: app_queue: Update Data of Queues (use queues as outbound calls container)
ASTERISK-19863: CallCounter not utilized by app_queue - members in busy state
ASTERISK-19864: Asterisk replying to Session progress with an Ack then an Invite
ASTERISK-19865: Forward a received 'answered elsewhere'
ASTERISK-19866: Display pause reasont at cli queue show queuename
ASTERISK-19867: asterisk fails to lower its priority when astcanary dies
ASTERISK-19868: How to enable ExtensionState Check with SIP real time
ASTERISK-19871: No translation path between various SILK sample rates
ASTERISK-19874: Deadlock when voicemail ODBC database fails to respond. Asterisk does not respond to any sip requests.
ASTERISK-19875: Behavior change in BLINDTRANSFER variable such that it is not available at the h extension
ASTERISK-19876: app_voicemail: make_email_file() sends emails with localized Date header
ASTERISK-19877: Queued Calls Remain In Queue When Phone In Queue Reboots
ASTERISK-19880: Can't transcode between SILK codecs
ASTERISK-19881: My Asterisk crashes multiple times daily
ASTERISK-19882: Asterisk fails to unsubscribe from PubSub nodes when using ejabberd
ASTERISK-19883: [patch] - RTP packet with Timestamp=0 on Multicast paging
ASTERISK-19886: MOH class set by mohinterpret does not show up in NoOp(${CHANNEL(musicclass)})
ASTERISK-19887: Pattern matching broken on Local channels (ast 1.8)
ASTERISK-19888: Choppy Audio from one client, great audio from two others, difference in RTP type number, why?
ASTERISK-19889: Asterisk crashes due to memory corruption
ASTERISK-19890: Crash occurs when using SIP realtime with SQL Server database, if a SIP client is un-registered
ASTERISK-19891: Realtime queue problem with joinempty option
ASTERISK-19892: If Asterisk sends a 481 to an initial INVITE that contained a to-tag, then Asterisk will not recognize the ensuing ACK
ASTERISK-19894: Asterisk ooh323 channl has voice lost during established calls - log notes: No Open LogicalChannels
ASTERISK-19896: safe_asterisk
ASTERISK-19898: OriginateResponse event is received with Reason 3, 'remote end is ringing', while remote end destination is unreachable
ASTERISK-19899: Confbridge user number announcement segfaults for number > 2
ASTERISK-19901: Asterisk 1.8.13.0 Blockers
ASTERISK-19902: Asterisk 10.5.0 Blockers
ASTERISK-19903: Remove AST_FLAG_ANSWERED_ELSEWHERE, duplicating the functionality of AST_CAUSE_ANSWERED_ELSEWHERE
ASTERISK-19905: Security Vulnerability: remotely exploitable crash in chan_skinny if client is disconnected when client is not in on-hook state
ASTERISK-19908: Add an ami function to refresh a voicemail box
ASTERISK-19911: echocan_mode not documented
ASTERISK-19912: [patch] Add ANI-2 / OLI reporting to chan_sip
ASTERISK-19914: Incorrect SIP cause to Asterisk cause mapping in chan_sip
ASTERISK-19915: CLIP - India
ASTERISK-19916: Asterisk gets exhausted of all the file resources
ASTERISK-19917: When WaitExten is called immediately on a new SIP call - DTMF fails to be detected with the RFC2833 and In-band methods, but succeeds with SIP INFO
ASTERISK-19918: MoH (Music on Hold) is stopped after call in a queue is terminated
ASTERISK-19919: Incorrect a=inactive when call changes from SIP_PAGE2_CALL_ONHOLD_INACTIVE to SIP_PAGE2_CALL_ONHOLD_ONEDIR
ASTERISK-19920: res_adsi module is loaded (or Asterisk thinks it is) despite no modules.conf, noload or autoload=no instructions
ASTERISK-19921: codec_dahdi: Wrong number of encoder/decoder channels.
ASTERISK-19922: AudioHook mutex errors when Local channel is optimized away
ASTERISK-19923: Asterisk crashing due to memory corruptions in chan_sip/voicemail
ASTERISK-19924: Setting the variable CHANNEL(tonezone) when using AMI command Originate seems to have no effect
ASTERISK-19925: Accountcode field in realtime does not work!
ASTERISK-19934: [patch] dialplan reload context
ASTERISK-19935: Patch to make app_system check if a command failed to execute due to permission denied.
ASTERISK-19936: Segmentation fault with extensions.conf not present, empty or everything commented out
ASTERISK-19937: Still not working - Failed to CANCEL a call in ringing state (SIP) in 1.8.9.2, 1.8.11, 1.8.12.2
ASTERISK-19939: Write test for the Asterisk Test Suite to cover subscribing for MWI in chan_sip
ASTERISK-19941: Crash in res_config_ldap when used with realtime extensions
ASTERISK-19942: SIP Session Timers do not honour refresher
ASTERISK-19943: Ref leak in app_mixmonitor, manager_mixmonitor
ASTERISK-19948: Asterisk 1.8 manager redirect command fails when redirecting multiple channels currently bridged together via dial command.
ASTERISK-19949: app_meetme unable to write frame to Local/XXX channel (stuck channel)
ASTERISK-19957: cdr_adaptative_odbc missing records
ASTERISK-19960: Incorrect data in queue_log, event TRANSFER, field data1
ASTERISK-19961: Completely silenced tone zone
ASTERISK-19962: Asterisk 1.8.12.0 can not play files in exten h
ASTERISK-19963: Add AccountCode field to the manager hangup event
ASTERISK-19965: Add IPv6 Support To Manager
ASTERISK-19966: Masquerade super test fails when timing source is timer_fd
ASTERISK-19968: TCP Session-Timers not dropping call
ASTERISK-19969: Enhance astobj2 to support other types of containers.
ASTERISK-19970: Add red-black tree container to astobj2.
ASTERISK-19971: Segfault in realtime_multi_ldap (possible invalid arguments)
ASTERISK-19974: AMI ORIGINATE - can't set CALLERID(num-pres) on outgoing call
ASTERISK-19977: CLONE - Asterisk res ldap
ASTERISK-19978: Gtalk Channel can't hangup on Android ICS 4.0.4
ASTERISK-19979: Request URI port inclusion inconsistency during outbound registration
ASTERISK-19981: Dial plan variables are limited to 4K (4096 bytes)
ASTERISK-19983: ConfBridge does not expose a mechanism to change the language on the Bridging channel, defaulting to 'en'
ASTERISK-19984: sip configuration with insecure
ASTERISK-19985: Bridge - not always returning to the right context,priority,extension as it should.
ASTERISK-19986: meetme - Error writing file for recording
ASTERISK-19987: Create chan_jingle2/res_xmpp Test Plan
ASTERISK-19989: Add XMPP support to Asterisk Test Suite
ASTERISK-19991: Memory leak in cel_pgsql
ASTERISK-19992: SIP re-INVITEs have no transaction timeout
ASTERISK-19994: app_voicemail should be able to decide on storage engines at runtime
ASTERISK-19995: Recording calls is strongly degraded when using RTP packetization of 60 ms (g729: 60)
ASTERISK-19996: Asterisk logs two CDR entries for a Local call.
ASTERISK-19997: Faulty asterisk sip registrations - cause incoming network buffer to rise